cup.c

FreeAMP(MP3播放)程序源代码-用来研究MP3解码

/*____________________________________________________________________________
	
	FreeAmp - The Free MP3 Player

        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
        Corp.  http://www.xingtech.com

	Portions Copyright (C) 1998-1999 EMusic.com

	This program is free software; you can redistribute it and/or modify
	it under the terms of the GNU General Public License as published by
	the Free Software Foundation; either version 2 of the License, or
	(at your option) any later version.

	This program is distributed in the hope that it will be useful,
	but WITHOUT ANY WARRANTY; without even the implied warranty of
	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
	GNU General Public License for more details.

	You should have received a copy of the GNU General Public License
	along with this program; if not, write to the Free Software
	Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
	
	$Id: cup.c,v 1.5 2000/10/13 14:29:02 ijr Exp $
____________________________________________________________________________*/

/****  cup.c  ***************************************************

MPEG audio decoder Layer I/II  mpeg1 and mpeg2
should be portable ANSI C, should be endian independent


mod  2/21/95 2/21/95  add bit skip, sb limiting

mods 11/15/95 for Layer I

******************************************************************/
/******************************************************************

       MPEG audio software decoder portable ANSI c.
       Decodes all Layer I/II to 16 bit linear pcm.
       Optional stereo to mono conversion.  Optional
       output sample rate conversion to half or quarter of
       native mpeg rate. dec8.c adds oupuut conversion features.

-------------------------------------
int audio_decode_init(MPEG *m, MPEG_HEAD *h, int framebytes_arg,
         int reduction_code, int transform_code, int convert_code,
         int freq_limit)

initilize decoder:
       return 0 = fail, not 0 = success

MPEG *m			input, mpeg structure for multiple streams
MPEG_HEAD *h    input, mpeg header info (returned by call to head_info)
framebytes      input, mpeg frame size (returned by call to head_info)
reduction_code  input, sample rate reduction code
                    0 = full rate
                    1 = half rate
                    2 = quarter rate

transform_code  input, ignored
convert_code    input, channel conversion
                  convert_code:  0 = two chan output
                                 1 = convert two chan to mono
                                 2 = convert two chan to left chan
                                 3 = convert two chan to right chan
freq_limit      input, limits bandwidth of pcm output to specified
                frequency.  Special use. Set to 24000 for normal use.


---------------------------------
void audio_decode_info( MPEG *m, DEC_INFO *info)

MPEG *m			input, mpeg structure for multiple streams
information return:
          Call after audio_decode_init.  See mhead.h for
          information returned in DEC_INFO structure.


---------------------------------
IN_OUT audio_decode(unsigned char *bs, void *pcmbuf)

decode one mpeg audio frame:
bs        input, mpeg bitstream, must start with
          sync word.  Caution: may read up to 3 bytes
          beyond end of frame.
pcmbuf    output, pcm samples.

IN_OUT structure returns:
          Number bytes conceptually removed from mpeg bitstream.
          Returns 0 if sync loss.
          Number bytes of pcm output.

*******************************************************************/


#include <stdlib.h>
#include <stdio.h>
#include <float.h>
#include <math.h>
#include "L3.h"
#include "mhead.h"		/* mpeg header structure */


#ifdef _MSC_VER
#pragma warning(disable: 4709)
#endif


/*-------------------------------------------------------
NOTE:  Decoder may read up to three bytes beyond end of
frame.  Calling application must ensure that this does
not cause a memory access violation (protection fault)
---------------------------------------------------------*/

/*====================================================================*/
/*----------------*/

/* Read Only */
static int look_joint[16] =
{				/* lookup stereo sb's by mode+ext */
   64, 64, 64, 64,		/* stereo */
   2 * 4, 2 * 8, 2 * 12, 2 * 16,	/* joint */
   64, 64, 64, 64,		/* dual */
   32, 32, 32, 32,		/* mono */
};	/* Okay to be global */

/* Read Only */
static int bat_bit_master[] =
{
   0, 5, 7, 9, 10, 12, 15, 18, 21, 24, 27, 30, 33, 36, 39, 42, 45, 48};


void sbt_mono(MPEG *m, float *sample, short *pcm, int n);
void sbt_dual(MPEG *m, float *sample, short *pcm, int n);


IN_OUT L2audio_decode(void *mv, unsigned char *bs, signed short *pcm);

/*======================================================================*/
/*======================================================================*/

/*------------- initialize bit getter -------------*/
static void load_init(MPEG *m, unsigned char *buf)
{
   m->cup.bs_ptr = buf;
   m->cup.bits = 0;
   m->cup.bitbuf = 0;
}
/*------------- get n bits from bitstream -------------*/
static long load(MPEG *m, int n)
{
   unsigned long x;

   if (m->cup.bits < n)
   {				/* refill bit buf if necessary */
      while (m->cup.bits <= 24)
      {
	 m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++;
	 m->cup.bits += 8;
      }
   }
   m->cup.bits -= n;
   x = m->cup.bitbuf >> m->cup.bits;
   m->cup.bitbuf -= x << m->cup.bits;
   return x;
}
/*------------- skip over n bits in bitstream -------------*/
static void skip(MPEG *m, int n)
{
   int k;

   if (m->cup.bits < n)
   {
      n -= m->cup.bits;
      k = n >> 3;
/*--- bytes = n/8 --*/
      m->cup.bs_ptr += k;
      n -= k << 3;
      m->cup.bitbuf = *m->cup.bs_ptr++;
      m->cup.bits = 8;
   }
   m->cup.bits -= n;
   m->cup.bitbuf -= (m->cup.bitbuf >> m->cup.bits) << m->cup.bits;
}
/*--------------------------------------------------------------*/
#define mac_load_check(n) if( m->cup.bits < (n) ) {                           \
          while( m->cup.bits <= 24 ) {               \
             m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++;  \
             m->cup.bits += 8;                       \
          }                                   \
   }
/*--------------------------------------------------------------*/
#define mac_load(n) ( m->cup.bits -= n,                    \
         m->cup.bitval = m->cup.bitbuf >> m->cup.bits,      \
         m->cup.bitbuf -= m->cup.bitval << m->cup.bits,     \
         m->cup.bitval )
/*======================================================================*/
static void unpack_ba(MPEG *m)
{
   int i, j, k;
   /* Read Only */
   static int nbit[4] =
   {4, 4, 3, 2};
   int nstereo;

   m->cup.bit_skip = 0;
   nstereo = m->cup.stereo_sb;
   k = 0;
   for (i = 0; i < 4; i++)
   {
      for (j = 0; j < m->cup.nbat[i]; j++, k++)
      {
	 mac_load_check(4);
	 m->cup.ballo[k] = m->cup.samp_dispatch[k] = m->cup.bat[i][mac_load(nbit[i])];
	 if (k >= m->cup.nsb_limit)
	    m->cup.bit_skip += bat_bit_master[m->cup.samp_dispatch[k]];
	 m->cup.c_value[k] = m->cup.look_c_value[m->cup.samp_dispatch[k]];
	 if (--nstereo < 0)
	 {
	    m->cup.ballo[k + 1] = m->cup.ballo[k];
	    m->cup.samp_dispatch[k] += 18;	/* flag as joint */
	    m->cup.samp_dispatch[k + 1] = m->cup.samp_dispatch[k];	/* flag for sf */
	    m->cup.c_value[k + 1] = m->cup.c_value[k];
	    k++;
	    j++;
	 }
      }
   }
   m->cup.samp_dispatch[m->cup.nsb_limit] = 37;	/* terminate the dispatcher with skip */
   m->cup.samp_dispatch[k] = 36;	/* terminate the dispatcher */

}
/*-------------------------------------------------------------------------*/
static void unpack_sfs(MPEG *m)	/* unpack scale factor selectors */
{
   int i;
   for (i = 0; i < m->cup.max_sb; i++)
   {
      mac_load_check(2);
      if (m->cup.ballo[i])
	 m->cup.sf_dispatch[i] = mac_load(2);
      else
	 m->cup.sf_dispatch[i] = 4;	/* no allo */
   }
   m->cup.sf_dispatch[i] = 5;		/* terminate dispatcher */
}
/*-------------------------------------------------------------------------*/
static void unpack_sf(MPEG *m)		/* unpack scale factor */
{				/* combine dequant and scale factors */